Systems and methods are disclosed herein for improving audio conferencing services. One aspect relates to processing audio content of a conference. A first audio signal is received from a first conference participant, and a start and an end of a first utterance by the first conference participant are detected from the first audio signal. A second audio signal is received from a second conference participant, and a start and an end of a second utterance by the second conference participant is detected from the second audio signal. The second conference participant is provided with at least a portion of the first utterance, wherein at least one of start time, start point, and duration is determined based at least in part on the start, end, or both, of the second utterance.

One example may include a method that includes receiving, at a server, a data set from one or more mobile devices located in a presentation space, combining the received data set with additional data to create a combined data set, creating a presentation signal based on the combined data set, subtracting a portion of one or more of the data set and the additional data set from the combined data set to create a modified presentation signal, forwarding the modified presentation signal to one or more of a display and a loudspeaker located in the presentation space, and playing the modified presentation signal via one or more of the loudspeaker and the display device.

One example may include a method that includes initiating an audio recording to capture audio data, comparing the audio data received from a microphone of a mobile device to an audio data range, determining whether the audio data is above an optimal level based on a result of the comparison, and queuing the audio data in an audio data queue when the audio data is above the optimal level.

One example may include a method that includes receiving, at a presentation server, an audio data signal from a mobile device located in a presentation space, identifying a mobile device identification characteristic of the mobile device based on the received audio data signal, determining a mobile device location via a location determination procedure, and playing the audio signal via a loudspeaker.

Systems and methods for controlling hearing devices are disclosed. Each hearing device is worn by a user, the hearing devices being communicatively interconnected with each other for data exchange to form a hearing device system. A method includes acquiring a sound signal of each user with a microphone of the hearing device of the user; determining from the sound signal a speaker signal, which indicates, whether the user is a speaking user; determining a priority score for each user from the speaker signal of the user; outputting sound signals from speaking users to the users via the hearing devices of the users, such that sound signals from speaking users with a priority score smaller than a threshold are muted.

A cloud-based video/audio conferencing system and method performs locale based aggregation of a full duplex media stream to organize multiple connections to a conference call that originate from the same physical location or a shared locale. The cloud-based video/audio conferencing system performs synchronization of the microphone and speaker audio signals of the same-locale connections. In this manner, a conference call may be held with multiple user devices making connections from the same physical location. User experience is enhanced by allowing each user in the same location to use his/her own individual devices to connect to the same conference call.

Systems and methods are provided for managing a conference call with multiple voice-enabled and voice-capturing devices, such as smart speakers. Reproduced, duplicate voice commands can cause unexpected results in a conference call. The voice commands can be determined to be received from the same conference call. A voice command for a particular voice-enabled device can be selected based on an energy level of an audio signal, event data, time data, and/or user identification.

To filter unwanted sounds from a conference call, a voice profile of a first user is generated based on a first voice signal captured by a media device during a first conference call. The voice profile may be generated by identifying a base frequency of the first voice signal and determining a plurality of voice characteristics, such as pitch, intonation, accent, loudness, and speech rate. These data may be stored in association with the first user. During a second conference call, a second voice signal captured by the media device is analyzed to determine, based on the voice profile of the first user, whether the second voice signal includes the voice of a second user. If so, the second voice signal is prevented from being transmitted into the conference call. A voice profile of the second user may be generated from the second voice signal for future use.

Systems and methods are disclosed herein for improving audio conferencing services. One aspect relates to processing audio content of a conference. A first audio signal is received from a first conference participant, and a start and an end of a first utterance by the first conference participant are detected from the first audio signal. A second audio signal is received from a second conference participant, and a start and an end of a second utterance by the second conference participant is detected from the second audio signal. The second conference participant is provided with at least a portion of the first utterance, wherein at least one of start time, start point, and duration is determined based at least in part on the start, end, or both, of the second utterance.

Systems and methods are provided for managing a conference call with multiple voice-enabled and voice-capturing devices, such as smart speakers. Reproduced, duplicate voice commands can cause unexpected results in a conference call. The voice commands can be determined to be received from the same conference call. A voice command for a particular voice-enabled device can be selected based on an energy level of an audio signal, event data, time data, and/or user identification.